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feat: AudioSocket channel, application, and ARI support.
This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
This commit is contained in:
Seán C McCord
Ben Ford
345
res/res_audiosocket.c
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345
res/res_audiosocket.c
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2019, CyCore Systems, Inc
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*
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* Seán C McCord <scm@cycoresys.com
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief AudioSocket support for Asterisk
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*
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* \author Seán C McCord <scm@cycoresys.com>
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*
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*/
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/*** MODULEINFO
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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#include "errno.h"
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#include <uuid/uuid.h>
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#include "asterisk/file.h"
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#include "asterisk/res_audiosocket.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/uuid.h"
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#include "asterisk/format_cache.h"
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#define MODULE_DESCRIPTION "AudioSocket support functions for Asterisk"
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#define MAX_CONNECT_TIMEOUT_MSEC 2000
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/*!
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* \internal
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* \brief Attempt to complete the audiosocket connection.
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*
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* \param server Url that we are trying to connect to.
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* \param addr Address that host was resolved to.
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* \param netsockfd File descriptor of socket.
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*
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* \retval 0 when connection is succesful.
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* \retval 1 when there is an error.
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*/
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static int handle_audiosocket_connection(const char *server,
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const struct ast_sockaddr addr, const int netsockfd)
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{
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struct pollfd pfds[1];
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int res, conresult;
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socklen_t reslen;
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reslen = sizeof(conresult);
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pfds[0].fd = netsockfd;
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pfds[0].events = POLLOUT;
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while ((res = ast_poll(pfds, 1, MAX_CONNECT_TIMEOUT_MSEC)) != 1) {
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if (errno != EINTR) {
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if (!res) {
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ast_log(LOG_WARNING, "AudioSocket connection to '%s' timed"
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"out after MAX_CONNECT_TIMEOUT_MSEC (%d) milliseconds.\n",
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server, MAX_CONNECT_TIMEOUT_MSEC);
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} else {
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ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", server,
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strerror(errno));
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}
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return -1;
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}
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}
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if (getsockopt(pfds[0].fd, SOL_SOCKET, SO_ERROR, &conresult, &reslen) < 0) {
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ast_log(LOG_WARNING, "Connection to %s failed with error: %s\n",
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ast_sockaddr_stringify(&addr), strerror(errno));
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return -1;
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}
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if (conresult) {
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ast_log(LOG_WARNING, "Connecting to '%s' failed for url '%s': %s\n",
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ast_sockaddr_stringify(&addr), server, strerror(conresult));
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return -1;
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}
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return 0;
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}
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const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
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{
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int s = -1;
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struct ast_sockaddr *addrs;
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int num_addrs = 0, i = 0;
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if (chan && ast_autoservice_start(chan) < 0) {
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ast_log(LOG_WARNING, "Failed to start autoservice for channel "
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"%s\n", ast_channel_name(chan));
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goto end;
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}
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if (ast_strlen_zero(server)) {
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ast_log(LOG_ERROR, "No AudioSocket server provided\n");
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goto end;
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}
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if (!(num_addrs = ast_sockaddr_resolve(&addrs, server, PARSE_PORT_REQUIRE,
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AST_AF_UNSPEC))) {
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ast_log(LOG_ERROR, "Failed to resolve AudioSocket service using %s - "
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"requires a valid hostname and port\n", server);
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goto end;
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}
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/* Connect to AudioSocket service */
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for (i = 0; i < num_addrs; i++) {
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if (!ast_sockaddr_port(&addrs[i])) {
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/* If there's no port, other addresses should have the
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* same problem. Stop here.
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*/
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ast_log(LOG_ERROR, "No port provided for %s\n",
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ast_sockaddr_stringify(&addrs[i]));
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s = -1;
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goto end;
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}
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if ((s = ast_socket_nonblock(addrs[i].ss.ss_family, SOCK_STREAM,
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IPPROTO_TCP)) < 0) {
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ast_log(LOG_WARNING, "Unable to create socket: %s\n", strerror(errno));
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continue;
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}
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if (ast_connect(s, &addrs[i]) && errno == EINPROGRESS) {
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if (handle_audiosocket_connection(server, addrs[i], s)) {
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close(s);
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continue;
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}
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} else {
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ast_log(LOG_ERROR, "Connection to %s failed with unexpected error: %s\n",
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ast_sockaddr_stringify(&addrs[i]), strerror(errno));
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close(s);
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s = -1;
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}
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break;
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}
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end:
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if (addrs) {
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ast_free(addrs);
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}
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if (chan && ast_autoservice_stop(chan) < 0) {
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ast_log(LOG_WARNING, "Failed to stop autoservice for channel %s\n",
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ast_channel_name(chan));
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close(s);
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return -1;
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}
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if (i == num_addrs) {
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ast_log(LOG_ERROR, "Failed to connect to AudioSocket service\n");
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close(s);
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return -1;
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}
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return s;
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}
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const int ast_audiosocket_init(const int svc, const char *id)
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{
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uuid_t uu;
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int ret = 0;
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uint8_t buf[3 + 16];
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if (ast_strlen_zero(id)) {
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ast_log(LOG_ERROR, "No UUID for AudioSocket\n");
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return -1;
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}
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if (uuid_parse(id, uu)) {
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ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", id);
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return -1;
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}
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buf[0] = 0x01;
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buf[1] = 0x00;
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buf[2] = 0x10;
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memcpy(buf + 3, uu, 16);
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if (write(svc, buf, 3 + 16) != 3 + 16) {
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ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
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ret = -1;
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}
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return ret;
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}
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const int ast_audiosocket_send_frame(const int svc, const struct ast_frame *f)
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{
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int ret = 0;
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uint8_t kind = 0x10; /* always 16-bit, 8kHz signed linear mono, for now */
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uint8_t *p;
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uint8_t buf[3 + f->datalen];
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p = buf;
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*(p++) = kind;
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*(p++) = f->datalen >> 8;
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*(p++) = f->datalen & 0xff;
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memcpy(p, f->data.ptr, f->datalen);
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if (write(svc, buf, 3 + f->datalen) != 3 + f->datalen) {
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ast_log(LOG_WARNING, "Failed to write data to AudioSocket\n");
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ret = -1;
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}
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return ret;
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}
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struct ast_frame *ast_audiosocket_receive_frame(const int svc)
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{
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int i = 0, n = 0, ret = 0, not_audio = 0;
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struct ast_frame f = {
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.frametype = AST_FRAME_VOICE,
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.subclass.format = ast_format_slin,
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.src = "AudioSocket",
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.mallocd = AST_MALLOCD_DATA,
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};
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uint8_t kind;
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uint8_t len_high;
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uint8_t len_low;
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uint16_t len = 0;
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uint8_t *data;
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n = read(svc, &kind, 1);
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if (n < 0 && errno == EAGAIN) {
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return &ast_null_frame;
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}
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if (n == 0) {
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return &ast_null_frame;
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}
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if (n != 1) {
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ast_log(LOG_WARNING, "Failed to read type header from AudioSocket\n");
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return NULL;
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}
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if (kind == 0x00) {
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/* AudioSocket ended by remote */
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return NULL;
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}
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if (kind != 0x10) {
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/* read but ignore non-audio message */
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ast_log(LOG_WARNING, "Received non-audio AudioSocket message\n");
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not_audio = 1;
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}
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n = read(svc, &len_high, 1);
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if (n != 1) {
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ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
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return NULL;
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}
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len += len_high * 256;
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n = read(svc, &len_low, 1);
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if (n != 1) {
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ast_log(LOG_WARNING, "Failed to read data length from AudioSocket\n");
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return NULL;
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}
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len += len_low;
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if (len < 1) {
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return &ast_null_frame;
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}
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data = ast_malloc(len);
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if (!data) {
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ast_log(LOG_ERROR, "Failed to allocate for data from AudioSocket\n");
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return NULL;
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}
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ret = 0;
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n = 0;
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i = 0;
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while (i < len) {
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n = read(svc, data + i, len - i);
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if (n < 0) {
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ast_log(LOG_ERROR, "Failed to read data from AudioSocket\n");
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ret = n;
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break;
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}
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if (n == 0) {
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ast_log(LOG_ERROR, "Insufficient data read from AudioSocket\n");
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ret = -1;
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break;
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}
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i += n;
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}
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if (ret != 0) {
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ast_free(data);
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return NULL;
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}
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if (not_audio) {
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ast_free(data);
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return &ast_null_frame;
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}
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f.data.ptr = data;
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f.datalen = len;
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f.samples = len / 2;
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/* The frame steals data, so it doesn't need to be freed here */
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return ast_frisolate(&f);
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}
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static int load_module(void)
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{
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ast_verb(1, "Loading AudioSocket Support module\n");
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
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{
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ast_verb(1, "Unloading AudioSocket Support module\n");
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER,
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"AudioSocket support",
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.support_level = AST_MODULE_SUPPORT_EXTENDED,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_CHANNEL_DEPEND,
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);
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